Linux-audio-user April 2014

linux-audio-user@lists.linuxaudio.org

101 participants
94 discussions

by Len Ovens

On Wed, 16 Apr 2014, david wrote: > On 04/16/2014 03:17 AM, Len Ovens wrote: >> I was talking about through air. I can only play as fast as I can hear >> and at thirty feet or so from the sound source, my instrument sounds >> delayed from the rest of the band because I hear their sound that much >> later and play my part that much later. I don't normally play at that >> distance, but ten feet is pretty standard. > > You're a few orders of magnitude better than me! I don't think I'd notice > that much difference. I used to play and try to adjust sound while practicing, so I had 50 feet or so of cable. By the time I got to the sound board it was getting hard to play. > I use headphone monitor. I find monitoring with headphones when playing bass is hard. I use a small amp tilted back about 10 feet ahead of me... so 13-14 feet to my ear I guess. Bass needs some space to work. My amp just gives me enough for me and the board can add whatever they want to the mains. I go a bit heavy on the high end for my monitor and the mains can roll that off for house sound. > That's my understanding, too. Things like staying away from USB mice and > keyboards, making sure the USB audio is the only device on that bus, yah? On that irq... same for pci(e) really even prioritizing shared irqs, lone irqs are still best. >> However, it looks like I can still get lots of MB with PCI slots in >> them. I will probaly do that. Hopefully with three PCI slots I can get >> one that is irq clean. > > My desktop has 2 PCIe slots; the other 4 are PCI. It also has 4 USB ports + > 2 more USB connector points on the mobo, 4 SATA connectors, an EIDE > connector, a floppy connector, plus the built-in audio and video and > Ethernet. I haven't checked to see what's sharing interrupts and what isn't. > I wouldn't be surprised if there aren't a bunch of shared interrupts! cat /proc/interrupts will tell you that quick enough. For some reason a lot of MB share irq16 with 3 or 4 things... worse one of those tings is often one of the PCI(e) slots. No need really as most modern MB have access to 48 plus irqs. My old board has 24 but 2-7 are unused (for hysterical reasons?) and 10 and 13 seem to be skipped too. I think 20 goes to my internal audio which I have turned off (AC97) so it doesn't show. I have noticed that on any of the MB I have checked or seen irq maps for that the internal audio always has it's own irq and quite often it is the highest one which in most systems these days has the most priority. I found this out because I had the card I was using for midi in the higher of the two slots and my audio below. I had trouble with xruns on the audio, but when I put the audio on the higher irq I had no more trouble. I know that in theory that shouldn't happen because there are two part to the irq drivers, a stub to answer the IRQ and save enough info to work and the other part that the os prioritizes and does all the work. So the os should be able to prioritize by the module name. I just know what I have found works best. I have heard the words "in a modern system" too many times. I think any system can do better audio if it is tuned/tweaked. -- Len Ovens www.ovenwerks.net

11 years, 10 months

[LAU] Interview about theater music produced with Ardour3

by Brett McCoy

I recently did an interview regarding the theater company I write music for. I am posting this because I produce all of my music with Linux audio (although Windows samplers like Kontakt are used, the real work I describe here is done with Linux). I also mention MuseScore and Ardour3, here, too :-) http://www.art-stream.org/content/flair-dramatic-music-great-scott (the singing your hear is me, I am a horrible singer but recorded scratch vocals for the actors, even Autotalent couldn't fix my warbling) -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi

11 years, 10 months

Re: [LAU] hardware - Intel CPUs

by Len Ovens

On Tue, 15 Apr 2014, James Mckernon wrote: > On Mon, Apr 14, 2014 at 2:06 AM, Len Ovens <len(a)ovenwerks.net> wrote: > I would realy like to stay away from having to use a USB or FW > audio IF. In fact I would like to be able to continue to use my > delta 66 for as long as I can before I spend more money :) The > > Thanks for the useful info in your post. Just to be clear on this part: are > you saying you don't want to switch to USB/FW solely because you want to > keep using your delta 66, or because you have some definite preference for > PCIe over USB/FW devices? If the latter, I wonder why? USB in audio is limited. Getting clear USB ports interupt wise is not easy. Audio can not be on a hub or share it's usb with anything else, but many new MB have no mouse or kb port so the USB is already being used for that much. The real reason though, is latency. With the pci the latency can be 1/4 what it can be in USB or FW. That is the lowest seeting jack for USB or FW is 64/2, but I can run the d66 at 16/2 with no problem on a well tuned system. This does make a difference for live work. I know that 64/2 seems like very good latency (it is) but remember that the card then adds another ms in each direction as well as the stage distances on top of that. That is the time it takes the sound to reach my ear after going through the computer as a processor and then through the air to my ear. Maybe that is still not worth worrying about... but even with 30 feet of cord and no digital delay, I can hear the delay from my playing to the sound reaching my ear. -- Len Ovens www.ovenwerks.net

11 years, 10 months

Re: [LAU] hardware - Intel CPUs

by Len Ovens

On Thu, 17 Apr 2014, david wrote: > On 04/17/2014 09:56 AM, Len Ovens wrote: >> >> >> On Wed, 16 Apr 2014, david wrote: >> >>> Then running your sound from stage to backhouse sound board back to >>> stage and hearing it through headphones would give no latency at all. >> >> Nothing I can hear. This is very important becasue if there was >> noticable delay then stage sound would be ahead of the mains... might >> give some interesting filtering from being out of phase. > > The mains are getting their signal via cable, so they're getting their > signal as quickly as the monitors on stage. The only change would be > physical distance from the different speakers, right? The difference I was thinking of is between a DIed guitar amp and mains as these and the bass amp (sometimes KB too) have traditionally been faced towards the audience. We also have some pretty loud vocalists who's unamplified voice can be heard in the house. -- Len Ovens www.ovenwerks.net

11 years, 10 months

[LAU] jack2 vs. jack1/zita vs. jack2+zita-a2j, performance differences

by Jonathan E. Brickman

Thought the below might be of interest to some. The last is the best :-) Hardware and clients identical, octo 4GHz, test load is one yoshimi with jack_keyboard driving by way of a passthrough mididings. ---------------------------- Item one is vanilla distro jack2, run like this: jack_control start jack_control ds alsa jack_control dps device hw:SB jack_control dps rate 48000 jack_control dps nperiods 2 jack_control dps period 128 jack_control dps midi-driver seq jack_control dps inchannels 2 jack_control dps outchannels 2 jack_control eps realtime true jack_control eps realtime-priority 50 jack_control eps clock-source 1 The above is as low in reported latency I could go, 5.33ms, using every optimization I know of via the Arch wiki and realtimeconfigquickscan and other sources. Running load-less, it gives me usage ranging 0.9 through 1.4%, usually hanging in at 1.3%. With the test load, it tends to sit at 10% and rise with stress. A few xruns every once in a while are inevitable, even when the test load is delivering only silence. Nice tone quality. ----------------------------- Item two is jack1-git, compiled with Zita libraries engaged, run like this: nohup schedtool -R -p 50 -e /usr/bin/jackd -A SB -R -c h -X alsa_midi -d dummy -r 48000 -p 32 & Reported latency is 2ms. Load-less, usage rating is 1.3% through 29%, usually hanging in at 1.4% or so. With the test load, it sits at 31.2% at silence. No xruns. But no actual sound came out :-) ----------------------------- Item three is jack1-git, Zita libraries present but not used, run like this: nohup schedtool -R -p 50 -e /usr/bin/jackd -R -c h -X alsa_midi -d dummy -r 48000 -p 32 & nohup schedtool -R -p 50 -e /usr/local/bin/zita-j2a -d hw:SB 2ms reported latency. Loadless usage is 20%. With test load, 25-33% at silence. No xruns. Distorted sound. ------------------------------ Item four is jack2 vanilla distro, run like this: jack_control start jack_control ds dummy jack_control dps rate 48000 jack_control dps nperiods 2 jack_control dps period 32 jack_control dps inchannels 2 jack_control dps outchannels 2 jack_control eps realtime true jack_control eps realtime-priority 50 jack_control eps clock-source 1 schedtool -R -p 50 -e /usr/bin/a2jmidid schedtool -R -p 50 -e /usr/local/bin/zita-j2a -d hw:SB 2ms reported latency. Loadless usage 3.8 to 4.3%. With triple test load (three Yoshimis which is actually SOP for this rig), 4.5-5.5% at silence. No xruns. Wonderful sound, and no xruns with that triple load being pushed. Will be starting rigorous testing at lower latencies soon, but I have an instrument which has to sing first :-) Jonathan E. Brickman Ponderworthy Music | jeb(a)ponderworthy.com | (785)233-9977 | http://ponderworthy.com

11 years, 10 months

[LAU] Anyone used a Firewire-to-USB3 or ESATA adapter?

by david

My church has a 16-channel Presonus Firewire device that we're not currently using because (1) my laptop has no Firewire connector (the dead old one did), and (2) our sound tech's laptop has Windows 7 on it and apparently there's no Windows 7 device driver for his Firewire PC Card adapter. So I thought, maybe a Firewire<->USB adaptor would work. All we used the Presonus for is recording individual channels. Now I just record 2 channels of essentially-mono output from our mixing board, but I very much miss the ability to clean up and EQ individual instruments and mix them better (the volunteer sound people who frequently run our mixing board don't exactly have the greatest ears, plus the acoustics in our rented location aren't the best). My laptop has USB2 and USB3, plus an ESATA port. I see Amazon has a selection: Firewire to USB http://www.amazon.com/s?ie=UTF8&page=1&rh=n%3A172282%2Ck%3Afirewire%20to%20… Amazon doesn't seem to show a Firewire-ESATA adaptor. For desktop folk, they list this PCIe card (2 external ESATA, 2 external Firewire800/1394B, 1 internal 1394B, runs TI XIO2213 chipset): http://www.amazon.com/NitroAV-Fusion-FireWire800-Professional-Adapter/dp/B0… Unfair: There's an Apple Thunderbolt to FireWire Adapter! I guess that will be enough to keep Firewire alive in the pro audio world. -- David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

11 years, 10 months

[LAU] M-Audio MTRACK

by Kaza Kore

Looking to buy a new USB audio interface in the next couple of days as am at a place where I will be staying for a little while (some of you may remember my previous post about the cards available on Amazon India as I'm currently travelling.) Well had a look at Ebay and come acros the M-Audio MTRACK. Sure it should be Linux compatible as it claims to be Class Compliant but I have a question about the headphone output which isn't answered specifically in the manual or FAQ. Can anybody confirm whether it is a duplicate of the Main output audio? Or if it is individually configurable at all? I suspect the former, but I have been happily surprised in the past, finding a headphone output to actually be a stereo output of its own, rather than a Master duplicate. Feel this very unlikely in the case but thought I would ask just in case anybody knows (or wants to warn me off for other reasons)... http://www.m-audio.com/images/global/manuals/M-Track_-_User_Guide_-_v1.1.pdf http://www.m-audio.com/products/en_us/MTrack.html

11 years, 10 months

[LAU] Fwd: Music made in Linux - A song from an hobbyst

by Carlo Ascani

2014-04-16 20:52 GMT+02:00 F. Silvain <silvain(a)freeshell.de>: > Did you use a real analogue drum machine or are those sounds from a software > synth? Also what tool did you use for the effect sound? The electronic drums are samples played by hydrogen and processed by plugins (calf mostly). On the second part the real drum mixes with those electronic samples. I used mostly calf plugins (eqs, comps and reverbs), and a couple of effects from guitarix (distorsion and tube screamer) > Thank you again. That's a pleasure to share things with you guys! > ... > > Ta-ta > ---- > Ffanci > * Internet: http://freeshell.de/~silvain -- Carlo Ascani | carlorat.me skype: carloratm -- Carlo Ascani | carlorat.me skype: carloratm

11 years, 10 months

[LAU] [LAA] Guitarix 0.29.0 released

by hermann meyer

The Guitarix developers proudly present Guitarix release 0.29.0 For the uninitiated, Guitarix is a tube amplifier simulation for jack, with effect modules and an additional stereo effect chain. Among with a couple changes in the source and some bug-fixes, this relase comes with a couple of new plugs (gx & LV2), were most of them comes from our new Development Member Fedor Uporov. The Simulation plugs been developed with the new Analog Circuit Simulation Toolkit by Andreas Degert. new plugs: * Record Mono/Stereo (gx) * JCM800PRE (Preamp simulation) (gx) * GCB_95 (WahWah simulation) (gx) * Duck Delay Mono/Stereo (gx / LV2) * Reverse Delay (gx) * Graphic EQ (gx / LV2) * Ring Modulator Mono/Stereo (gx) * Plate Reverb (gx) * Panorama Enhancer (gx) * Bass Enhancer (gx) * BarkGraphicEQ (24band LV2) Please refer to our project page for more information: http://guitarix.sourceforge.net/ Download Site: http://sourceforge.net/projects/guitarix/ Forum: http://guitarix.sourceforge.net/forum/ Please consider visiting our forum or leaving a message on guitarix-developer(a)lists.sourceforge.net <mailto:guitarix-developer@lists.sourceforge.net>

11 years, 10 months

[LAU] Music made in Linux - A song from an hobbyst

by Carlo Ascani

Hi, I have just uploaded a song made entirely in Linux. The links: on bandcamp - https://lotfi.bandcamp.com/track/fulci direct download - http://carlorat.me/download/fulci.flac I have uploaded the full Ardour 3 project on bitbucket, it contains all the stems without processing. https://bitbucket.org/carloratm/fulci/src Please, feel free to open it and remix! All audio data is under a Creative Commons By Share-Alike license (see COPYING) And now, just a couple of words about the song with some credits: - the original idea was composed using Renoise - the speech you can hear is Lucio Fulci, an Italian director - the bass is played and recorded by Lorenzo Cecchi on linux - the analogue drums was played by me and recorded on linux by Lorenzo Cecchi - the piano was played by frate (http://amanitaphalloides.bandcamp.com/) - the mixing has been done by me, using Ardour 3. I am in no way a professionist neither an expert. I did it just for fun, I hope you enoy it. -- Carlo Ascani | carlorat.me skype: carloratm

11 years, 10 months

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Linux-audio-user April 2014